17 Commits

Author SHA1 Message Date
Niels Möller
280632b440 Delete unneeded forward declares of RtpReceiver
Bug: webrtc:7135
Change-Id: I1ca8537248ed5c87f8913263c680e0a5a5544130
Reviewed-on: https://webrtc-review.googlesource.com/94506
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24353}
2018-08-21 09:30:02 +00:00
Kári Tristan Helgason
798ee75d23 Always disable RED when ULPFEC is disabled.
This prevents a lot of unnecessary processing taking place when we are
not using FEC.

This CL also removes the FieldTrial that was used to disable ulpfec, as it's no longer used.

Bug: webrtc:9514
Change-Id: I8285b933f71eea971f5932cd19833455a42c8639
Reviewed-on: https://webrtc-review.googlesource.com/87848
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23952}
2018-07-12 13:01:20 +00:00
Danil Chapovalov
d264df587f Replace rtc::Optional with absl::optional in modules/rtp_rtcp
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-15 09:53:35 +00:00
Erik Språng
eeaa8f929c Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
Bug: webrtc:9271
Change-Id: Ic7415830588bef9d87bab92943460207890dada6
Reviewed-on: https://webrtc-review.googlesource.com/76960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 11:22:56 +00:00
Erik Språng
566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.

Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.

Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 15:31:27 +00:00
Steve Anton
296a0ce4c7 Add MID sending to RTPSender
This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.

Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
2018-03-23 01:50:45 +00:00
Niels Möller
2e1d784956 Delete the VideoCodec::plName string.
It holds the same information as codecType, but in different format.

Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
2018-03-06 11:17:41 +00:00
Erik Språng
7b52f102ef Don't write pacer exit timestamp without pacer
And allow populating network2 timestamp if we want to preserve pacer
timestamp.

Bug: webrtc:8853
Change-Id: I895d5ce8a9cca8ceeec3bf08e2eff02bf3b2f5fd
Reviewed-on: https://webrtc-review.googlesource.com/48640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21937}
2018-02-07 14:45:43 +00:00
Jiawei Ou
3587b8302a Make RTCP report interval configurable
Bug: webrtc:8789
Change-Id: I79c9132123c946b030ed79c647b4329e81d6e6ae
Reviewed-on: https://webrtc-review.googlesource.com/43201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21837}
2018-02-01 10:12:11 +00:00
Danil Chapovalov
eb0edd832a Narrow interface PacketRouter use to send Remb and TransportFeedback
This allows to use RtcpTransceiver implementation instead of RtpRtcp.
No functional changes.

Bug: webrtc:8239
Change-Id: I3c5bd23ff2136eb844e85b567b70380fc2a65929
Reviewed-on: https://webrtc-review.googlesource.com/33005
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21298}
2017-12-15 15:58:17 +00:00
Danil Chapovalov
1de4b62955 Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb
in particular change bitrate type to int64_t to follow style guide.

With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter

Bug: webrtc:8239
Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba
Reviewed-on: https://webrtc-review.googlesource.com/32302
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21250}
2017-12-13 14:40:01 +00:00
Danil Chapovalov
651707bdf0 Remove deprecated SetREMB functions from RtpRtcp
Bug: None
Change-Id: I8b299d8e83d99fc2d074df876c95ca2680226efa
Reviewed-on: https://webrtc-review.googlesource.com/22061
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20676}
2017-11-14 18:21:46 +00:00
Danil Chapovalov
51e21aaa7a Simplify RtpRtcp interface for REMB
Remove REMB accessor as used for debug checks only.
Merge SetRembData and SetRembStatus(true) eliminating 
state 'remb can be send, but no data available yet'

Bug: None
Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf
Reviewed-on: https://webrtc-review.googlesource.com/7983
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20240}
2017-10-11 11:09:39 +00:00
nisse
435472542a Delete deprecated metod RtpRtcp::SetMaxTransferUnit.
Deprecated since cl https://codereview.webrtc.org/2589743002

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/3006413002
Cr-Commit-Position: refs/heads/master@{#19878}
2017-09-18 07:37:37 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00